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(Auto)update libjingle 75924589-> 75925673

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@7251 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-09-19 20:18:10 +00:00
parent 0c9fe72b21
commit c98f217c65
2 changed files with 8 additions and 7 deletions
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8
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -2884,7 +2884,7 @@ bool WebRtcVideoMediaChannel::SetStartSendBandwidth(int bps) {
}
// On success, SetSendCodec() will reset |send_start_bitrate_| to |bps/1000|,
// by calling MaybeChangeBitrates. That method will also clamp the
// by calling SanitizeBitrates. That method will also clamp the
// start bitrate between min and max, consistent with the override behavior
// in SetMaxSendBandwidth.
webrtc::VideoCodec new_codec = *send_codec_;
@ -3659,7 +3659,7 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
<< "for ssrc: " << ssrc << ".";
} else {
StreamParams* send_params = send_channel->stream_params();
MaybeChangeBitrates(channel_id, &target_codec);
SanitizeBitrates(channel_id, &target_codec);
webrtc::VideoCodec current_codec;
if (!engine()->vie()->codec()->GetSendCodec(channel_id, current_codec)) {
// Compare against existing configured send codec.
@ -3946,7 +3946,7 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
vie_codec.codecSpecific.VP8.denoisingOn = enable_denoising;
vie_codec.codecSpecific.VP8.frameDroppingOn = vp8_frame_dropping;
}
MaybeChangeBitrates(channel_id, &vie_codec);
SanitizeBitrates(channel_id, &vie_codec);
if (engine()->vie()->codec()->SetSendCodec(channel_id, vie_codec) != 0) {
LOG_RTCERR1(SetSendCodec, channel_id);
@ -3984,7 +3984,7 @@ bool WebRtcVideoMediaChannel::MaybeResetVieSendCodec(
return true;
}
void WebRtcVideoMediaChannel::MaybeChangeBitrates(
void WebRtcVideoMediaChannel::SanitizeBitrates(
int channel_id, webrtc::VideoCodec* codec) {
codec->minBitrate = GetBitrate(codec->minBitrate, kMinVideoBitrate);
codec->startBitrate = GetBitrate(codec->startBitrate, kStartVideoBitrate);

7
talk/media/webrtc/webrtcvideoengine.h
View File

@ -310,6 +310,10 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
virtual int SendPacket(int channel, const void* data, int len);
virtual int SendRTCPPacket(int channel, const void* data, int len);
// Checks the current bitrate estimate and modifies the bitrates
// accordingly, including converting kAutoBandwidth to the correct defaults.
virtual void SanitizeBitrates(
int channel_id, webrtc::VideoCodec* video_codec);
virtual void LogSendCodecChange(const std::string& reason);
bool SetPrimaryAndRtxSsrcs(
int channel_id, int idx, uint32 primary_ssrc,
@ -361,9 +365,6 @@ class WebRtcVideoMediaChannel : public rtc::MessageHandler,
bool MaybeResetVieSendCodec(WebRtcVideoChannelSendInfo* send_channel,
int new_width, int new_height, bool is_screencast,
bool* reset);
// Checks the current bitrate estimate and modifies the bitrates
// accordingly, including converting kAutoBandwidth to the correct defaults.
void MaybeChangeBitrates(int channel_id, webrtc::VideoCodec* video_codec);
// Helper function for starting the sending of media on all channels or
// |channel_id|. Note that these two function do not change |sending_|.
bool StartSend();